November 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v3] ASoC: rt5645: Wait for 400msec before concluding on value of RT5645_VENDOR_ID2
by Alex Deucher 09 Nov '17

09 Nov '17

From: Akshu Agrawal <akshu.agrawal(a)amd.com> Minimum time required between power On of codec and read of RT5645_VENDOR_ID2 is 400msec. We should wait that long before reading the value. TEST=Cold boot the device and check for sound device. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> Signed-off-by: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Alex Deucher <alexander.deucher(a)amd.com> --- v2: rework the patch based on mailing list discussion. Just wait before reading the register. v3: Drop the extra read sound/soc/codecs/rt5645.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 23cc2cb8393f..5f24df4fae8e 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -55,6 +55,8 @@ MODULE_PARM_DESC(quirk, "RT5645 pdata quirk override"); #define RT5645_HWEQ_NUM 57 +#define TIME_TO_POWER_MS 400 + static const struct regmap_range_cfg rt5645_ranges[] = { { .name = "PR", @@ -3784,6 +3786,12 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, ret); return ret; } + + /* + * Read after 400msec, as it is the interval required between + * read and power On. + */ + msleep(TIME_TO_POWER_MS); regmap_read(regmap, RT5645_VENDOR_ID2, &val); switch (val) { -- 2.13.6

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[alsa-devel] [PATCH] ALSA: hda - Fix memory leak on snd_hdac_device_init error
by Guneshwor Singh 09 Nov '17

09 Nov '17

From: "Subhransu S. Prusty" <subhransu.s.prusty(a)intel.com> Free codec vendor name on snd_hdac_device_init error. Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Signed-off-by: Guneshwor Singh <guneshwor.o.singh(a)intel.com> --- sound/hda/hdac_device.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 19deb306facb..467b949544e2 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -111,6 +111,8 @@ int snd_hdac_device_init(struct hdac_device *codec, struct hdac_bus *bus, return 0; error: + kfree(codec->vendor_name); + put_device(&codec->dev); return err; } -- 2.15.0

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[alsa-devel] [PATCH v2] ASoC: rt5663: Delay and retry reading rt5663 ID register
by Oder Chiou 09 Nov '17

09 Nov '17

In the probe, the codec may not be ready for I2C reading or there are some glitches on the i2c line. So if the i2c reading value is incorrect, it will read again after delay. This issue is similar the patch https://patchwork.kernel.org/patch/9681421/. In current project, these 2 devices were connected to the same i2c line, and they met the same problem. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5663.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index fcd1643..5985d8e 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -3429,7 +3429,16 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, ret); return ret; } - regmap_read(regmap, RT5663_VENDOR_ID_2, &val); + + ret = regmap_read(regmap, RT5663_VENDOR_ID_2, &val); + if (ret || (val != RT5663_DEVICE_ID_2 && val != RT5663_DEVICE_ID_1)) { + dev_err(&i2c->dev, + "Device with ID register %#x is not rt5663," + " retry one time.\n", val); + msleep(100); + regmap_read(regmap, RT5663_VENDOR_ID_2, &val); + } + switch (val) { case RT5663_DEVICE_ID_2: rt5663->regmap = devm_regmap_init_i2c(i2c, &rt5663_v2_regmap); -- 2.7.4

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[alsa-devel] [PATCH v2 0/3] Fixes for AMD Stoney ACP audio
by Alex Deucher 09 Nov '17

09 Nov '17

This patch set is just a couple fixes for the Audio CoProcessor (ACP) on AMD Stoney platforms and a small general fix for the rt5645 codec driver. The entire patch set can also be viewed here: https://cgit.freedesktop.org/~agd5f/linux/log/?h=sound-for-next-stoney2 Thanks! Alex v2: - Add a fix for a 64 bit divide in hw_ptr patch - rework rt5645 patch as per Mark's comments Akshu Agrawal (2): ASoC: AMD: Make the driver name consistent across files ASoC: rt5645: Wait for 400msec before concluding on value of RT5645_VENDOR_ID2 Guenter Roeck (1): FIXUP: FROMLIST: ASoC: amd: Report accurate hw_ptr during dma sound/soc/amd/Makefile | 4 ++-- sound/soc/amd/acp-pcm-dma.c | 9 +++++---- sound/soc/codecs/rt5645.c | 9 +++++++++ 3 files changed, 16 insertions(+), 6 deletions(-) -- 2.13.6

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[alsa-devel] [PATCH] ALSA: hda - fix headset mic problem for Dell machines with alc274
by Hui Wang 09 Nov '17

09 Nov '17

Confirmed with Kailang of Realtek, the pin 0x19 is for Headset Mic, and the pin 0x1a is for Headphone Mic, he suggested to apply ALC269_FIXUP_DELL1_MIC_NO_PRESENCE to fix this problem. And we verified applying this FIXUP can fix this problem. Cc: <stable(a)vger.kernel.org> Cc: Kailang Yang <kailang(a)realtek.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9b77d43..db1a376 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6568,6 +6568,11 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x14, 0x90170110}, {0x1b, 0x90a70130}, {0x21, 0x03211020}), + SND_HDA_PIN_QUIRK(0x10ec0274, 0x1028, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0xb7a60130}, + {0x13, 0xb8a61140}, + {0x16, 0x90170110}, + {0x21, 0x04211020}), SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4, {0x12, 0x90a60130}, {0x14, 0x90170110}, -- 2.7.4

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[alsa-devel] Applied "ASoC: cs42l56: Fix reset GPIO name in example DT binding" to the asoc tree
by Mark Brown 08 Nov '17

08 Nov '17

The patch ASoC: cs42l56: Fix reset GPIO name in example DT binding has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 8adc430603d67e76a0f8491df21654f691acda62 Mon Sep 17 00:00:00 2001 From: "Andrew F. Davis" <afd(a)ti.com> Date: Wed, 8 Nov 2017 15:24:59 -0600 Subject: [PATCH] ASoC: cs42l56: Fix reset GPIO name in example DT binding The binding states the reset GPIO property shall be named "cirrus,gpio-nreset" and this is what the driver looks for, but the example uses "gpio-reset". Fix this here. Fixes: 3bb40619aca8 ("ASoC: cs42l56: bindings: sound: Add bindings for CS42L56 CODEC") Signed-off-by: Andrew F. Davis <afd(a)ti.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/devicetree/bindings/sound/cs42l56.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/cs42l56.txt b/Documentation/devicetree/bindings/sound/cs42l56.txt index 4feb0eb27ea4..4ba520a28ae8 100644 --- a/Documentation/devicetree/bindings/sound/cs42l56.txt +++ b/Documentation/devicetree/bindings/sound/cs42l56.txt @@ -55,7 +55,7 @@ Example: codec: codec@4b { compatible = "cirrus,cs42l56"; reg = <0x4b>; - gpio-reset = <&gpio 10 0>; + cirrus,gpio-nreset = <&gpio 10 0>; cirrus,chgfreq-divisor = <0x05>; cirrus.ain1_ref_cfg; cirrus,micbias-lvl = <5>; -- 2.15.0

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[alsa-devel] [PATCH] ASoC: amd: use do_div rather than 64 bit division to fix 32 bit builds
by Alex Deucher 08 Nov '17

08 Nov '17

From: Guenter Roeck <groeck(a)chromium.org> ERROR: "__aeabi_uldivmod" [sound/soc/amd/snd-soc-acp-pcm.ko] undefined! 64-bit divides require special operations to avoid build errors on 32-bit systems. [Reword the commit message to make it clearer - Alex] fixes: 61add8147942 (ASoC: amd: Report accurate hw_ptr during dma) Signed-off-by: Guenter Roeck <groeck(a)chromium.org> Reviewed-on: https://chromium-review.googlesource.com/678919 Reviewed-by: Jason Clinton <jclinton(a)chromium.org> (cherry picked from commit 7ca726e80f21abdbaed9a5a70def1c33a26f8533) Reviewed-on: https://chromium-review.googlesource.com/681618 Signed-off-by: Alex Deucher <alexander.deucher(a)amd.com> --- sound/soc/amd/acp-pcm-dma.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 13d040a4d26f..ef7e98ad960c 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -856,12 +856,11 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { if (bytescount > rtd->renderbytescount) bytescount = bytescount - rtd->renderbytescount; - pos = bytescount % buffersize; } else { if (bytescount > rtd->capturebytescount) bytescount = bytescount - rtd->capturebytescount; - pos = bytescount % buffersize; } + pos = do_div(bytescount, buffersize); return bytes_to_frames(runtime, pos); } -- 2.13.6

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[alsa-devel] [PATCH 0/3] ASoC: sun8i-codec: A33 codec fixes
by Maxime Ripard 08 Nov '17

08 Nov '17

Hi, The commit 043b8daa578f ("ASoC: sun4i-i2s: Update global enable with bitmask") has broken the A33 codec since it's been merged as part as 4.14-rc1. The reason has been that while that commit was obviously right, one of its side effect was to clear the bit that would set the I2S controller in slave mode, effectively forcing it to act as master. This was actually working by accident since the codec also had its condition on whether to act as a master or a slave backward, meaning that even though we were setting the i2s controller as slave and the codec as master, we were ending up with the exact opposite. The commit mentionned above (rightfully) broke that combination, and we ended up with the two devices in slave mode, which obviously didn't work. The way to fix that is in two parts. The first one is obviously to make the codec act as its proper role. However, that's not sufficient because some logic was missing in the initial driver that was merged to act as such. Indeed the BCLK divider was never programmed, meaning that we would generate a BCLK running at the PLL rate, which is way too fast. The second patch addresses that. The final fix is here to address an issue that has been there for quite some time too and would invert the two channels. It appears that the codec and I2S drivers don't share the same polarity for the LRCK signal. The fix is quite obvious and is to invert whatever value has been programmed in ALSA. All these commits should go as fix in 4.14 if possible, as without them the audio is completely broken. Let me know what you think, Maxime Maxime Ripard (3): ASoC: sun8i-codec: Invert Master / Slave condition ASoC: sun8i-codec: Set the BCLK divider ASoC: sun8i-codec: Fix left and right channels inversion sound/soc/sunxi/sun8i-codec.c | 63 +++++++++++++++++++++++++++++++++++++++---- 1 file changed, 58 insertions(+), 5 deletions(-) -- 2.14.3

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[alsa-devel] [PATCH 0/9] ASoC: prepare for Component conversion - step 2
by Kuninori Morimoto 08 Nov '17

08 Nov '17

Hi Mark These are resend of one of "replace component" patch series. I had posted it as below. Subject: Prepare to covert Codec/Platform into Component Date: Fri, 01 Sep 2017 14:18:11 +0900 Because it is big patch-set, to avoid sending patch bomb, I will repost it as small amount of patch set. To review from big picture, please check above patch-set. I will continue to resend remainings of it if this small patch-set were accepted Kuninori Morimoto (9): ASoC: snd_soc_component_driver has snd_pcm_ops ASoC: snd_soc_component_driver has snd_compr_ops ASoC: snd_soc_component_driver has pmdown_time ASoC: snd_soc_component_driver has endianness ASoC: snd_soc_component_driver has non_legacy_dai_naming ASoC: add snd_soc_component_read32 ASoC: add snd_soc_component_xxx_bias_level() ASoC: add snd_soc_component_cache_sync() ASoC: add snd_soc_dapm_kcontrol_component() include/sound/soc.h | 78 ++++++++ sound/soc/soc-compress.c | 461 +++++++++++++++++++++++++++++++++++++++++++---- sound/soc/soc-core.c | 83 +++++---- sound/soc/soc-io.c | 14 ++ sound/soc/soc-pcm.c | 378 ++++++++++++++++++++++++++++++++++++-- 5 files changed, 929 insertions(+), 85 deletions(-) -- 1.9.1

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[alsa-devel] Applied "ASoC: add snd_soc_component_xxx_bias_level()" to the asoc tree
by Mark Brown 08 Nov '17

08 Nov '17

The patch ASoC: add snd_soc_component_xxx_bias_level() has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 10e079d9874802c35dc3f51af922531e5423a6bf Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Mon, 6 Nov 2017 01:48:37 +0000 Subject: [PATCH] ASoC: add snd_soc_component_xxx_bias_level() snd_soc_codec_xxx_bias_level() (= for Codec) will be removed soon. This patch Component version of it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc.h | 45 +++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 45 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 747588273c96..f69b13c4a641 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1439,6 +1439,21 @@ static inline void snd_soc_codec_init_bias_level(struct snd_soc_codec *codec, snd_soc_dapm_init_bias_level(snd_soc_codec_get_dapm(codec), level); } +/** + * snd_soc_component_init_bias_level() - Initialize COMPONENT DAPM bias level + * @component: The COMPONENT for which to initialize the DAPM bias level + * @level: The DAPM level to initialize to + * + * Initializes the COMPONENT DAPM bias level. See snd_soc_dapm_init_bias_level(). + */ +static inline void +snd_soc_component_init_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + snd_soc_dapm_init_bias_level( + snd_soc_component_get_dapm(component), level); +} + /** * snd_soc_dapm_get_bias_level() - Get current CODEC DAPM bias level * @codec: The CODEC for which to get the DAPM bias level @@ -1451,6 +1466,19 @@ static inline enum snd_soc_bias_level snd_soc_codec_get_bias_level( return snd_soc_dapm_get_bias_level(snd_soc_codec_get_dapm(codec)); } +/** + * snd_soc_component_get_bias_level() - Get current COMPONENT DAPM bias level + * @component: The COMPONENT for which to get the DAPM bias level + * + * Returns: The current DAPM bias level of the COMPONENT. + */ +static inline enum snd_soc_bias_level +snd_soc_component_get_bias_level(struct snd_soc_component *component) +{ + return snd_soc_dapm_get_bias_level( + snd_soc_component_get_dapm(component)); +} + /** * snd_soc_codec_force_bias_level() - Set the CODEC DAPM bias level * @codec: The CODEC for which to set the level @@ -1466,6 +1494,23 @@ static inline int snd_soc_codec_force_bias_level(struct snd_soc_codec *codec, level); } +/** + * snd_soc_component_force_bias_level() - Set the COMPONENT DAPM bias level + * @component: The COMPONENT for which to set the level + * @level: The level to set to + * + * Forces the COMPONENT bias level to a specific state. See + * snd_soc_dapm_force_bias_level(). + */ +static inline int +snd_soc_component_force_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + return snd_soc_dapm_force_bias_level( + snd_soc_component_get_dapm(component), + level); +} + /** * snd_soc_dapm_kcontrol_codec() - Returns the codec associated to a kcontrol * @kcontrol: The kcontrol -- 2.15.0

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